Re: [asterisk-users] Free help
At 11:58 PM 10/18/2007, you wrote: I could write you a script to wash your car. You'd just need some kind of interface to do the mechanical part of the work. I have a script to wash a car so you don't have to write one: http://www.lazaino.com/application.html Sorry, couldn't resist. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and wall displays/reader boards
I know of a call centre that bought a cheap projector for that purpose. PaulH On Thu, 2007-10-18 at 23:28 -0700, o o wrote: Has anyone used an LED wall display with asterisk? I have a customer who has an ancient telecorp system that drives an LED wall display. It shows the number of agents signed in, calls in queue, hold time, etc. It also sounds an alarm if the hold time exceeds a set value. I'm looking to use asterisk to replace the telecorp system. I know it can do all the CDR and historical data, but I haven't found anything on this. The current display is currently connected via serial (rj-11) but I would be open to getting a newer board with IP connectivity. thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Live Conference about asterisk and voip: reminder 12:30 PM EDT Friday
As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Howto get origin IP address from SIP call reliably
Roger Schreiter wrote: What is a reliable way to read the real IP address of the origin of a SIP call? Maybe SIPCHANINFO(peerip) or SIPCHANINFO(recvip)? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring Groups
Rob Schall wrote: Here's what I'm looking to do exten = 10,1,Dial(SIP/1000SIP/1001,15,wW) exten = 10,2,Voicemail(u1000) This should ring both phones and they should keep ringing for the alloted time before moving on. However, it appears that if one of the phones is Busy, it returns with a busy and moves on without really ringing the second phone. Short of checking if the channels are available or using a queue, is there a way to ignore the return value and just make it ring for 10 seconds and then move on to the second step? Any Suggestions? It should work the way you expect it to work. We would really have to see the CLI output of the failure. Also remove the ,wW while testing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut()
On Fri, 2007-10-19 at 07:22 -0700, bilal ghayyad wrote: My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension Use the TIMEOUT() function like this: exten = 123,n,Set(TIMEOUT(response)=5) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
Or your could use a touch screen with Flash Operator Panel. Just a suggestion out of left field. -Original Message- From: Russell Brown [mailto:[EMAIL PROTECTED] Sent: Friday, October 19, 2007 1:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Receptionists Phone suggestions? (Not Snom370) Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details below if you/re interested). I've verified this problem with Snom who's response is that the receptionist should answer all of the incoming calls before trying to do a transfer - That's just Bonkers! So... any suggestions? Details of Snom 370 problem for the record: Snom370 gets a Call (Call A). Snom370 answers Call A. Call A wants to be transferred to Phone C. Snom370 has another call ringing (Call B). Snom370 presses HOLD button gets Dialtone. Call A is on Hold, Call B still ringing. Snom370 Dials Phone C (Call C). Snom370 talks to Call C. Snom370 presses TRANSFER. The display shows: CallA CallB The soft keys now show and . Pressing them does nothing. When the TRANSFER button is pressed again, CallA is connected to CallB (the original caller is now talking to the previously unanswered party) not what one wanted to happen! It's not difficult to see why my client is throwing their toys out of the pram and I'm going to have to replace the Snoms at my expense :-( -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best USB Handset and Softphone Combination
I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with this sort of deployment? It would be very nice if there was a central way of provisioning the phones. All machines are fairly new (newer than two years), they have very strict policies on downloads and streaming. Thanks in advance. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ResponseTimeOut()
Hi List; My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup To what this related? About my extensions.conf file, I set priorityjumpin = yes and I set autofallthrough = no (and I am sure it is not related to the problem with ResponseTimeout application). Any help? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Live Conference about asterisk and voip: reminder 12:30 PM EDT Friday
for those of you who have not joined the conference call yet, I highly recommend it. there is always several interesting tidbits that will help you in your * implementations... see you at 12:30p today! daveC randulo wrote: As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Glare on Incoming Calls
C F wrote: How on earth does this prevent Glare? Or even reduce it? I think he was providing his configuration in case there WAS a change he could make to reduce it. The only thing we could do was an option because our incoming lines were arranged in a hunt group. We made sure that we dial out working down the group. So the phone company starts with line one, then line two, etc., we start with line three, and then two... By using the Dial(ZAP/G1/blah) syntax. The capital G searches the zap channel group in reverse. If you don't have a hunt group from the phone company, this probably won't make a bit of difference to you. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2: Incoming calls answered prematurely?
Eric ManxPower Wieling wrote: The remote server is where your problem is. Thanks for the reply but I can call the extension in question normally and it works fine. The problem is that the IAX trunk appears to be answering before it knows if the physical destination is available or not. I have read through every option I can find on IAX and elsewhere and I can't see how this functionality can be changed or influenced. Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Glare on Incoming Calls
How on earth does this prevent Glare? Or even reduce it?
On 10/19/07, Gustavo Gonzalez [EMAIL PROTECTED] wrote:
How I change my configuration to reduce this issue. I have this setting on
my zapata.conf
signalling=fxs_ks
group=1
callgroup=1
pickupgroup=1
channel=1
signalling=fxs_ks
group=2
callgroup=1
pickupgroup=1
channel=2;
singalling=fxs_ks
group=3
callgroup=1
pickupgroup=1
channel=3;
singalling=fxs_ks
group=4
callgroup=1
pickupgroup=1
channel=4
and for outbound calls I have this context on my extensions.conf
[out-callb]
exten = 44,1,Set(LANGUAGE()=es)
exten = 44,n,ChanIsAvail(Zap/g1Zap/g2Zap/g3Zap/g4)
exten = 44,n,GotoIf($[${AVAILCHAN} = ]?4:6)
exten = 44,n,Congestion
exten = 44,n,Hangup
exten = 44,n,Playback,ggestion/varios/moment
exten = 44,n,SetMusicOnhold(dialtone)
exten = 44,n,Set(TIMEOUT(response)=10)
exten = 44,n,Set(TIMEOUT(digit)=5)
exten = 44,n,WaitExten(25|m(dialtone))
Date: Thu, 18 Oct 2007 17:07:03 -0400
From: C F [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Incoming calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Glare that's what it's called, if the number you advertise as your
business number is zap/1 then use zap/G1 to dial out, otherwise use
zap/g1 to dial out. This will reduce but not eliminate the problem.
On 10/18/07, Gustavo Gonzalez [EMAIL PROTECTED] wrote:
Hello I have a question about incoming calls on TDM400P cards. I want to
know why an incoming call appear in a sorpresive way on a phone that I
pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
channels )that are free. I have four lines connected to my TDM400P card
and
when I get a free Zap channel to call I hear the voice of a people on the
other side from an incomming call, I think that asterisk bridge my free
channel with incomming calls but how do this?Thanks for any idea.
Alejandro González
Grupo Gestión
4384-0660
www.grupo-gestion.com.ar
[EMAIL PROTECTED]
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[asterisk-users] Hide passwords in SIP.conf
Hi guys, There is other way instead plain text to define passwords in sip.conf ? In register, peers and extensions ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to H323 translator
It should be automatic. As long as you have a dial plan destination for the H.323 endpoint, it does not matter what the transport and protocol is. That's handled transparently by its various channels. You will have to configure the SIP and H.323 settings for the channel drivers, of course, but aside from that, should be pretty simple. On Fri, 19 Oct 2007, bilal ghayyad wrote: Hi All; If I installed H.323 on asterisk, and the caller phone was SIP endpoint while I need to route the call for a destination via an H.323 trunk, so Asterisk will do that SIP to H.323 translation automatically or I have to do also a configuration to SIP to H.323 translation? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using register = to let Asterisk register to another softswitch via SIP
The same way you do it with IAX2, pretty much. http://www.voip-info.org/wiki-Asterisk+config+sip.conf On Fri, 19 Oct 2007, bilal ghayyad wrote: Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk to register firstly then I can route my calls to that SIP trunk. In IAX2, we use the register = , so what shall we do in Asterisk? And how its format will be (if we will use register)? Or what is the solution? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
On 10/19/07, Per Jessen [EMAIL PROTECTED] wrote: Per Jessen wrote: [EMAIL PROTECTED] wrote: Did you set NAT Keep Alive Enable: = Yes for the line in question in the SPA's configuration? Uh, no, not specifically and I'm guessing it's not set by default? The SPA921 config has a NAT Keep Alive Intvl which is set to 15 by default, which I'm taking to mean it has NAT keep alives enabled. No, look under the Line 1 or Line 2 tab ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2: Incoming calls answered prematurely [RESOLVED]
Eric ManxPower Wieling wrote:
Voicemail will answer the line. 10 seconds is a pretty short timeout.
What you need to do is copy the CLI output of your failed calls from
BOTH servers and put them in this thread. Then we can SEE what Asterisk
is ACTUALLY doing.
Thanks for making me think :-)
My colleague was in this evening and we have sorted it out now.
I'm not quite sure how I was getting the results I did earlier as he
thinks his Asterisk server had stopped running...
Anyway - we now have worked out a little solution which seems to work well.
At my end, the macro responsible for dialling sets the callerid(name) so
we know the call comes from a user on the IVR selection, and it sets the
callerid(number) to the current context so we can see which business the
caller the wanted to get to.
At my partner's end, he uses a simple gotoif() function in his extension
context to test for an IVR call (via the callerid(name)) and then will
not go to local voicemail but simply hangup after 15 seconds.
If the call is from one of internal extensions, the callerid(name) is
not set to IVR so he deals with that as a normal call and after the
timeout, it goes to his local voicemail.
Thanks again for your help.
Here's some of the stuff just for reference if anyone else is
interested. If anyone has any questions please ask.
[main_menu]
; Dialplan for all unknown number callers
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=5) ; Max time between digits
exten = s,n,Set(TIMEOUT(response)=15) ; Max time to wait
exten = s,n,Wait(1)
exten = s,n,Background(welcome-to-bell-lord)
exten = s,n(resume),Background(press-3-for-tolc) ; Short dialogues,
easy to change
exten = s,n,Background(press-4-for-fondoo) ; rather than one long sentence
exten = s,n,Background(press-5-for-arrowtees) ; which might need to be
changed
exten = s,n,Background(press-6-for-gen-enq) ; frequently.
exten = s,n,WaitExten()
exten = 3,1,Goto(tolc,s,1) ; Dial 3 For The Open Learning Centre
exten = 4,1,Goto(fondoo,s,1) ; Dial 4 for Fondoo Internet
exten = 5,1,Goto(arrowtees,s,1) ; Dial 5 for ArrowTees
exten = 6,1,Goto(gen_enq,s,1) ; For all other enquiries press 6
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(resume)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup()
[tolc]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT}) ; Calls the
belllord Macro with the channel(s) to dial and the current context
(voicemail)
[fondoo]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT})
[arrowtees]
exten = s,1,Macro(belllord,${ALANL},${CONTEXT})
[gen_enq]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT})
; Call with Macro(belllord,channel,vmbox)
[macro-belllord] ; Uses macro and DIALSTATUS for local devices
exten = s,1,Set(CALLERID(all)=IVR ${ARG2})
exten = s,n,Dial(${ARG1},10,tr)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED],u) ; business is the
voicemail context, ${ARG2} is the context from which this call came
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED],b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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[asterisk-users] Linksys 941/942 configuration guide
Does anyone have this guide and be willing to share it with me? Thank in advance? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
We switched to T1(PRI) for high bandwidth voice quality, echo
I am using TE212P(which is a dual span Echo Chancellor hardware DTMF). I
have only one PRI connection from PSTN, but I implemented this 6 months agao
when there were no single span cards.
Sangoma just came with one in April, but I didnt wanted to go with that bcos I
havent seen review that the drivers are old but the card is great.
Now when I have a nice setup of PRI with 95 SIP extension to Asterisk. I
recently got A101D(which has Echo cancellor hardware DTMF) for my standby
asterisk.
Bot of these with their current drivers work great for Echo Voice Quality.
But my system(config) had a big issue with DTMF detection, which means when
someone calls main line then trys to punch my extension(123) the asterisk
think its 112 dials that person or a wrong # like 111 which is not an
extension. SO I had to resolvbe this with Digium by enabling hardware DTMF 6
months ago from software DTMF(I am not sure wthere this was asterisk issue of
DTMF, anyways I enabled hardware DTMF in Digium card it worked fine.
But now the new Sangoma card which I bough for backup didnt have the drivers
compatible to enabled the hardware DTMF. SO had songoma give me a custom drive
for their hardwrae DTMF they did within 20-25 days it works.
But you wouldnt find that driver sin Sangoma site, bcos they are still working
on them(for me they fixed for my model-- A101D)
So in my view both are great unless they work. Atleast I have been using
Digium TE212P for 6 months.
Also note your Network QoS is also important, we have seperate switches to
avoid QoS it depends uto organisation wish funding.
Also the type of Desktop VoIP phones you have.
I think I have said lot, let me know if this was helpful or I was just barking
... ha ha ha...
--
Deepak
Michael J. Liberatore [EMAIL PROTECTED] wrote: Hi all, i have been
using asterisk for a few years but i am about to do my first t1 setup. After
terrible quality issues between two business locations, we have decided to
purchase a point to point t1 from the local phone co. The internet is too
crappy, too much lag, queing and jitter. Most calls were dropped.
I was about to order two cisco routers with csu cards and remembered our
wonderful asterisk supports direct t1. I remembered digium and sangoma both
make these cards.
After some problems with a digium fxo card, i just ordered a sangoma a200
with echo cancellation. I was also leaning towards getting the single t1
sangoma card that is $499 from voip supply. But i know digium also makes
one. I was wondering if the digium card works better or much easier with
asterisk? The digium description says you can split the t1 for voice and data
which sounds nice since i will only be using probably 4 channels max of the
t1. Does the sangoma card also do this? I noticed the sangoma card has a 5
year warranty which is nice since i have had multiple digium fxo cards die.
Is there any other reason to get or the other?
Thank you all for your help. I am hoping this opens up a whole new world in
asterisk for me.
-Mike
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Re: [asterisk-users] Good, affordable IAX hardphones?
We use: http://www.ngnsky.com/product_info.php?cPath=21products_id=50 when we have the remote extension blues. It works quite well for us and the phone isn't that bad. On 10/19/07, Vincent [EMAIL PROTECTED] wrote: Hi SIP is such a pain to use when NAT is involved that I'm willing to buy an IAX hardphone for someone who works remotely over the Net and needs to get calls from our Asterisk server, itself behind a NAT. Are there good, affordable IAX phones you would recommend? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP usage with Asterisk
Yehavi Bourvine +972-8-9489444 wrote: In any case, I'll try this week to upgrade to 1.4.6 version and then add IMAP support and inform what happens. There have been _many_ IMAP related fixes sine 1.4.6. Please try the latest version, 1.4.13, instead. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hide passwords in SIP.conf
Frederico, Take a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret This is the only way I know of. -- Alex On Fri, 19 Oct 2007, Frederico Madeira wrote: Hi guys, There is other way instead plain text to define passwords in sip.conf ? In register, peers and extensions ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP to H323 translator
Hi All; If I installed H.323 on asterisk, and the caller phone was SIP endpoint while I need to route the call for a destination via an H.323 trunk, so Asterisk will do that SIP to H.323 translation automatically or I have to do also a configuration to SIP to H.323 translation? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I emulate SIP presence for an extension?
Ade Vickers wrote: Is it possible in Asterisk 1.4.x to issue a dialplan command which will set a phantom SIP extension to busy for as long as a caller is in the spam trap, back to idle when they finally give up hang up? http://www.asterisk.org/node/48325 http://www.asterisk.org/node/48360 Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extensions.conf for basic IVR?
Hello I've never built an IVR before, so I was wondering if someone could share some code from their extensions.conf that would perform some of thoses steps: 1. When a call comes in from the TDM FXO port, answer 2. If no CID, play message No CID available. Please type the number where you wish to be called back. Loop until OK or remote party hung up 3. When CID is available, play main menu : 1 for sales, 2 for support, 3 for any other question 4. Play If you wish, you can leave a message to explain what problem you met. When done, save message as WAV, and play message Your message will be sent to the department in charge. Thank you, and go back to main menu 5. Move WAV file to web server's directory where a PHP script lists available messages 6. Send an e-mail to the group involved, eg. [EMAIL PROTECTED], [EMAIL PROTECTED] , including a the caller's CID and a link to the WAV file Thanks for any tip. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I emulate SIP presence for an extension?
I recently implemented a simple spam trap extension for telemarketers - once identified as a telemarketer (usually they ask to speak to the person in charge of recruiting/website/purchasing/etc.), I simply offer to put them through to the person in question, dump them on a special extension which plays music for 15 seconds, then 1.5s silence, then a please wait, we're trying to put you through message; repeat until they give up waiting. I'm using a Grandstream GXP2000 phone, so I've got 7 presence lights; of which I'm only using a couple at the moment. Is it possible in Asterisk 1.4.x to issue a dialplan command which will set a phantom SIP extension to busy for as long as a caller is in the spam trap, back to idle when they finally give up hang up? The basic reason is twofold: 1) I want to see just how long they're willing to wait, and 2) For a sense of personal amusement (yes, I am a bad man) :) Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.15.1/1078 - Release Date: 18/10/2007 17:47 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hide passwords in SIP.conf
Frederico Madeira wrote: Hi guys, There is other way instead plain text to define passwords in sip.conf ? In register, peers and extensions ? Thanks. Depending on how your asterisk server is setup to run, if you chmod /etc/asterisk as 750 and the files underneath as 640, then only the user and group owner can read (+ only owner user can write). Others will not even see the existence of the directory or files... My server runs as user asterisk and group asterisk. Alan. -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
On 10/19/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... Not sure what to tell you but certainly it works without problems in CentOS/RHEL SuSE Linux. About the cards personally I like the sangoma cards. As you can see they have a better warranty than the digium cards. Also I feel they aren't as tied to a platform (Asterisk) as the Digium cards are. And some people claim some Digium cards have IRQ issues or problems with certain big-name server components (mainboards mainly) of which I haven't heard similar complaints for the Sangoma cards. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sorta OT: Bounty for Click to Call plugin for IE
On 10/17/07, Michael Graves [EMAIL PROTECTED] wrote: I'm in process of transitioning a number of offices to a hosted virtual pbx from Junction Networks. It's a combination of OpenSER and Asterisk. They have a nice click-to-call extension for Firefox, but I need the equivalent for IE so that it can work with our CRM system. Junction told me that they have a bounty on offer for this if someone's interested in doing the work. Would the availability of the Firefox code make it easier to do an ActiveX implementation? Can you use .call files I have an approx 1kb PHP script that can be used for click to call ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FollowMe recorded name filename variable?
Hmm.. I think it should be cleaning it up post-call already. If not, please open a bug on Mantis as that sounds like a bug. On 10/19/07, Anthony Messina [EMAIL PROTECTED] wrote: Is there a variable for the filename that is created by the FollowMe application when a is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, affordable IAX hardphones?
On Fri, 19 Oct 2007 14:16:40 -0700, Charles Alvis [EMAIL PROTECTED] wrote: http://www.ngnsky.com/product_info.php?cPath=21products_id=50 Thanks. I'll check it out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poll: Asterisk IMAP feedback (was: Is anyonesuccessfully using IMAP storage)
We tried with MS Exchange but couldn't get it to work (MS Exchange doesn't support a master account). CP From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Thursday, October 18, 2007 11:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Poll: Asterisk IMAP feedback (was: Is anyonesuccessfully using IMAP storage) Hello, Are you using Asterisk 1.4 ? If positive, are you then successfully using IMAP storage ? Your input would be very valuable to decide if rewite of IMAP storage could be considered as bug fix (non one uses IMAP now) or as a new feature (many use IMAP storage today). So please, take a few seconds to reply as up to now (4 answers), successful IMAP user share = 0% ! Regards PS: If someone has a more effective way to gather user feedback, do not hesitate to tell. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom ip330/ip501 second ethernet port
I Just wanted to add something here, Having separate VLAN does nothing in terms of QOS. In fact having a computer feeding from phone make more sense because phone will untag packets coming from PC. and after that its all about your switch how to prioritize packets. Unless there is a way in your switch to prioritize one Vlan over another Vlan, ( i guess it depends on your manufacture, i think Cisco does that and also uses CDP to discover phones) Having different Vlans is not your answer. the most you get is less broadcast. On 10/19/07, David Gomillion [EMAIL PROTECTED] wrote: On 10/19/07, Kevin Smith [EMAIL PROTECTED] wrote: Robert McNaught wrote: Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being directly related to asterisk, but since Polycom phones are used a lot with asterisk, it seems a good place to post ;-) Robert McNaught Hi Robert, While I'm not sure how our network compares with yours, we run about twenty 601 phones along with our office workstations (some stations are without a phone). Each station with a phone is connected with the other Ethernet port on the phone so we have one drop to each station. The phones are on a separate VLAN from the rest of the network as well. From the user end, I have not had a report of any problems with the connections, call quality, etc. I would say give it a shot, maybe with a larger network that could change, but for a small office like I'm in charge of, it is working just fine. Kevin We have a medium-sized network (120 polycoms of various persuasions, and 80 workstations), and we haven't had any real problems with phones ruining QoS. We have the phones on separate VLANs than the workstations. Actually, every switch has 4 VLANs defined: 2 voice, 2 data, so no VLAN has more than about 12 devices (about because sometimes we have to put a pocket switch in a room where the people want to add yet another computer). The echo from SIP to SIP with people using cheap headsets has affected us far more than any problems with PCs trying to suck the bandwidth. If I remember correctly, recent firmwares on the Polycom phones pretty much do the right thing, giving priority to the phone traffic. To summarize: works OK for us. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
Well this is the bug I am having with the make install of 1.4.5.1: http://bugs.digium.com/view.php?id=10156 Even though I got it to install ztcfg -vvv still says 1.4.4 also. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, October 19, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] First Time T1 Questions [EMAIL PROTECTED] wrote: On 10/19/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... Not sure what to tell you but certainly it works without problems in CentOS/RHEL SuSE Linux. About the cards personally I like the sangoma cards. As you can see they have a better warranty than the digium cards. Also I feel they aren't as tied to a platform (Asterisk) as the Digium cards are. And some people claim some Digium cards have IRQ issues or problems with certain big-name server components (mainboards mainly) of which I haven't heard similar complaints for the Sangoma cards. I know I've said this time and time again, but just for the purpose that this will be archived somewhere on the net, there should not be any more problems related to interrupts and specific servers. If there are, *please* let me know so that we can fix it. We have spent much of the last year or so getting rid of these problems, and we are very much committed to having 100% compatibility, and getting rid of our former reputation of having IRQ/motherboard problems. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys 941/942 configuration guide
Please see: http://spc.pifiu.com under SPA Phone Admin guide On 10/19/07, Bruce Komito [EMAIL PROTECTED] wrote: Does anyone have this guide and be willing to share it with me? Thank in advance? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
[EMAIL PROTECTED] wrote: On 10/19/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... Not sure what to tell you but certainly it works without problems in CentOS/RHEL SuSE Linux. About the cards personally I like the sangoma cards. As you can see they have a better warranty than the digium cards. Also I feel they aren't as tied to a platform (Asterisk) as the Digium cards are. And some people claim some Digium cards have IRQ issues or problems with certain big-name server components (mainboards mainly) of which I haven't heard similar complaints for the Sangoma cards. I know I've said this time and time again, but just for the purpose that this will be archived somewhere on the net, there should not be any more problems related to interrupts and specific servers. If there are, *please* let me know so that we can fix it. We have spent much of the last year or so getting rid of these problems, and we are very much committed to having 100% compatibility, and getting rid of our former reputation of having IRQ/motherboard problems. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
I hope 2 things need to be clear.
1) One call per line, needs to be set on the VoIP.
2)We user Polycom 501 for all Desktop Polycom 601 for reception.
http://media.polycom.com/usa/en/products/voice/soundpoint_ip/601/demo/index.html
OK, what I mean by one call per line
-- Polycom of SIP Phones usually comes with 3,6 etc line display for extensions.
-- And each line display can accept/call/hold total of 8 active phone calls per
line. This will cause problem if all is on the same line feed.
--So one needs to accept only one call per line in the VoIP phones config file.
I am not sure how ur line feeds are setup. I just wanted to let u know that
there can be aproblem with transfer if u have multiple calls comming on same
line display.
Or, may be I am wrong in understanding ur email.
--
Deepak
Russell Brown [EMAIL PROTECTED] wrote:
Does anyone have any suggestions for a decent receptionists phone?
Aastra? Grandstream?
Something with (potentially) lots of BLFs, large(ish) screen, headset
and most importantly the ability to transfer calls?
I've installed five Snom 370s that seemed ideal but my client is very
very unhappy as the Snom 370 can't transfer a call correctly if there's
another call coming in (details below if you/re interested). I've
verified this problem with Snom who's response is that the receptionist
should answer all of the incoming calls before trying to do a transfer -
That's just Bonkers!
So... any suggestions?
Details of Snom 370 problem for the record:
Snom370 gets a Call (Call A).
Snom370 answers Call A. Call A wants to be transferred to Phone C.
Snom370 has another call ringing (Call B).
Snom370 presses HOLD button gets Dialtone. Call A is on Hold, Call B
still ringing.
Snom370 Dials Phone C (Call C).
Snom370 talks to Call C.
Snom370 presses TRANSFER.
The display shows:
CallA
CallB
The soft keys now show and . Pressing them does nothing.
When the TRANSFER button is pressed again, CallA is connected to CallB
(the original caller is now talking to the previously unanswered party)
not what one wanted to happen!
It's not difficult to see why my client is throwing their toys out of
the pram and I'm going to have to replace the Snoms at my expense :-(
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Re: [asterisk-users] IAX2: Incoming calls answered prematurely?
Voicemail will answer the line. 10 seconds is a pretty short timeout.
What you need to do is copy the CLI output of your failed calls from
BOTH servers and put them in this thread. Then we can SEE what Asterisk
is ACTUALLY doing.
Alan Lord wrote:
Eric ManxPower Wieling wrote:
Alan Lord wrote:
Eric ManxPower Wieling wrote:
The remote server is where your problem is.
Thanks for the reply but I can call the extension in question normally
and it works fine. The problem is that the IAX trunk appears to be
answering before it knows if the physical destination is available or
not. I have read through every option I can find on IAX and elsewhere
and I can't see how this functionality can be changed or influenced.
How do you know that the far end is not answering and then providing an
ringing tone. Asterisk does not magically answer IAX calls. Playback
and Background as well as other apps will answer the line unless told
not to.
When I tried this test today, I know the far end wasn't answering
because my colleague, his computer and his SIP phone were not there. So
there is no way that that call should have been answered.
His extension definition is:
[internal]
exten=201,1,Dial(${ALANB},10)
exten=201,2,VoiceMail(u201)
exten=201,3,Hangup()
The call was cleared down almost as soon as it was answered so I am
unclear as to why this occurred.
Thanks
Alan
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[asterisk-users] Using register = to let Asterisk register to another softswitch via SIP
Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk to register firstly then I can route my calls to that SIP trunk. In IAX2, we use the register = , so what shall we do in Asterisk? And how its format will be (if we will use register)? Or what is the solution? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FollowMe recorded name filename variable?
Is there a variable for the filename that is created by the FollowMe application when a is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good, affordable IAX hardphones?
Hi SIP is such a pain to use when NAT is involved that I'm willing to buy an IAX hardphone for someone who works remotely over the Net and needs to get calls from our Asterisk server, itself behind a NAT. Are there good, affordable IAX phones you would recommend? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
In addition to my below question, i was wondering if anyone had a problem with installing zaptel on debian sarge. its a udev problem, make install thinks i am running udev, but when i fix the makefile to be like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so something is not right... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Friday, October 19, 2007 1:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] First Time T1 Questions Hi all, i have been using asterisk for a few years but i am about to do my first t1 setup. After terrible quality issues between two business locations, we have decided to purchase a point to point t1 from the local phone co. The internet is too crappy, too much lag, queing and jitter. Most calls were dropped. I was about to order two cisco routers with csu cards and remembered our wonderful asterisk supports direct t1. I remembered digium and sangoma both make these cards. After some problems with a digium fxo card, i just ordered a sangoma a200 with echo cancellation. I was also leaning towards getting the single t1 sangoma card that is $499 from voip supply. But i know digium also makes one. I was wondering if the digium card works better or much easier with asterisk? The digium description says you can split the t1 for voice and data which sounds nice since i will only be using probably 4 channels max of the t1. Does the sangoma card also do this? I noticed the sangoma card has a 5 year warranty which is nice since i have had multiple digium fxo cards die. Is there any other reason to get or the other? Thank you all for your help. I am hoping this opens up a whole new world in asterisk for me. -Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
It's just FOP which works well. Dependent on the quality of touch screen obviously. I haven't spend any time with FOP using Touch screens myself but I'm sure others here have. There was a thread a few days ago that got into it a bit. -Original Message- From: Mike Clark [mailto:[EMAIL PROTECTED] Sent: Friday, October 19, 2007 8:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370) shadowym wrote: Or your could use a touch screen with Flash Operator Panel. Just a suggestion out of left field. snipped a bunch shadowym: Do you have a specific setup w/touchscreen that you have deployed and that works well? Thanks, Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2: Incoming calls answered prematurely?
Alan Lord wrote: Eric ManxPower Wieling wrote: The remote server is where your problem is. Thanks for the reply but I can call the extension in question normally and it works fine. The problem is that the IAX trunk appears to be answering before it knows if the physical destination is available or not. I have read through every option I can find on IAX and elsewhere and I can't see how this functionality can be changed or influenced. How do you know that the far end is not answering and then providing an ringing tone. Asterisk does not magically answer IAX calls. Playback and Background as well as other apps will answer the line unless told not to. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Glare on Incoming Calls
Mojo with Horan Company, LLC wrote: C F wrote: How on earth does this prevent Glare? Or even reduce it? I think he was providing his configuration in case there WAS a change he could make to reduce it. The only thing we could do was an option because our incoming lines were arranged in a hunt group. We made sure that we dial out working down the group. So the phone company starts with line one, then line two, etc., we start with line three, and then two... By using the Dial(ZAP/G1/blah) syntax. The capital G searches the zap channel group in reverse. If you don't have a hunt group from the phone company, this probably won't make a bit of difference to you. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users They only way to eliminate a glare condition is to have your phone company convert you lines to ground start. Jonn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
Trunk and backport. IMHO that is the way to go. Philipp Kempgen wrote: Philipp Kempgen wrote: Steve Murphy wrote: But, in 1.4, I really can't add a new CDR field and call it a 'bug fix'. It really is a 'new', 'enhanced' sort of thing. So, this kind of change will have to go into trunk at the moment. Sad but true. I guess it couldn't go in even if there was a config option defaulting to off (i.e. old-style behavior)? Maybe it could be made available in the event on the manager interface without being classified as a new sort of thing. Just thinking out loud. Regards, Philipp Kempgen -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom ip330/ip501 second ethernet port
On 10/19/07, Kevin Smith [EMAIL PROTECTED] wrote: Robert McNaught wrote: Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being directly related to asterisk, but since Polycom phones are used a lot with asterisk, it seems a good place to post ;-) Robert McNaught Hi Robert, While I'm not sure how our network compares with yours, we run about twenty 601 phones along with our office workstations (some stations are without a phone). Each station with a phone is connected with the other Ethernet port on the phone so we have one drop to each station. The phones are on a separate VLAN from the rest of the network as well. From the user end, I have not had a report of any problems with the connections, call quality, etc. I would say give it a shot, maybe with a larger network that could change, but for a small office like I'm in charge of, it is working just fine. Kevin We have a medium-sized network (120 polycoms of various persuasions, and 80 workstations), and we haven't had any real problems with phones ruining QoS. We have the phones on separate VLANs than the workstations. Actually, every switch has 4 VLANs defined: 2 voice, 2 data, so no VLAN has more than about 12 devices (about because sometimes we have to put a pocket switch in a room where the people want to add yet another computer). The echo from SIP to SIP with people using cheap headsets has affected us far more than any problems with PCs trying to suck the bandwidth. If I remember correctly, recent firmwares on the Polycom phones pretty much do the right thing, giving priority to the phone traffic. To summarize: works OK for us. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile and Asterisk 1.2 ?
Hi, just noticed chan_mobile, which looks like it will do exactly as I need. http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels However seems it is only for latest 1.4 but there is a mention of a backport for 1.2 http://www.sigsegv.cx/sip-9.html Anybody using this with something like 1.2.18?? Care to share how you compiled it. Many thanks Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom ip330/ip501 second ethernet port
Hi Robert, While I'm not sure how our network compares with yours, we run about twenty 601 phones along with our office workstations (some stations are without a phone). Each station with a phone is connected with the other Ethernet port on the phone so we have one drop to each station. The phones are on a separate VLAN from the rest of the network as well. From the user end, I have not had a report of any problems with the connections, call quality, etc. I would say give it a shot, maybe with a larger network that could change, but for a small office like I'm in charge of, it is working just fine. Kevin Robert McNaught wrote: Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being directly related to asterisk, but since Polycom phones are used a lot with asterisk, it seems a good place to post ;-) Robert McNaught ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] DIDX Receives Digium Innovation Award
Philipp Kempgen wrote: Steve Totaro wrote: I am using Thunderbird 2.0.0.5. If using Outlook, I think the time is correct. Does MS have a different attitude towards timezones? :) Sorry. I forgot that they don't read RFCs. ;) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] DIDX Receives Digium Innovation Award
Steve Totaro wrote: I am using Thunderbird 2.0.0.5. If using Outlook, I think the time is correct. Does MS have a different attitude towards timezones? :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
Philipp Kempgen wrote: Steve Murphy wrote: But, in 1.4, I really can't add a new CDR field and call it a 'bug fix'. It really is a 'new', 'enhanced' sort of thing. So, this kind of change will have to go into trunk at the moment. Sad but true. I guess it couldn't go in even if there was a config option defaulting to off (i.e. old-style behavior)? Maybe it could be made available in the event on the manager interface without being classified as a new sort of thing. Just thinking out loud. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
Steve Murphy wrote: On Wed, 2007-10-17 at 02:41 +0200, Philipp Kempgen wrote: Steve Murphy wrote: (Sorry for quoting so much but I need to keep the context.) For instance, if Zap/52 dials Zap/51, 52 --- dials talks --- 51 who hookflashes and dials Zap/50, 51 --- dials --- 50 and 51 hangs up, leaving 52 and 50 to talk away, 52 --- talks --- 50 we should get 1 cdr that records the call from 52 to 51, which would last until the hookflash; No doubt about it. and a second CDR that would be from 51 to 50, which would start at either chan 50/51 channel creation time, or even at hookflash time, have an answer time when 50 picked up, and last until either 50 or 52 hang up. Right. But why should it start at 50-51 channel creation time? That way you would think (by looking at the CDRs) that 51 talked to 50 for longer than they did. I'd prefer hookflash time. Well, the start time isn't as important as the answer time; because your billing times from answer to end. True if billing was the only thing CDRs are good for. The time from start to answer is how much time it took to dial, wait, and have the call answered... which people usually don't pay as much attention to. Right. But nonetheless the value that gets stored should be as accurate as possible. Or else you could just store a random value because nobody cares about it anyway. ;) 50's channel creation time will be when 50 picked up the phone to answer the call from 51. 51's channel creation time will be when 51 picked up the phone to answer the call from 52. If we use 51's channel creation time as the start time, it would be possible to see that 52's conversation with 51 and 51's with 50, overlap. It may not help much, but it's a hint that 52 was there. Need to think about it for a while. How about splitting the src into rsp (who's responsible for the call, i.e. who should pay the bill) and src (who was involved in the audio bridge)? Example: rsp src dst duration billsec 5252 51 130 120 5151 50105 5152 50 3610 3600 Actually, I've been thinking about this; adding a CDR field to record the responsible party for a call is a good way to handle these situations. But, in 1.4, I really can't add a new CDR field and call it a 'bug fix'. It really is a 'new', 'enhanced' sort of thing. So, this kind of change will have to go into trunk at the moment. Sad but true. I guess it couldn't go in even if there was a config option defaulting to off (i.e. old-style behavior)? Since I can't add the new field into 1.4, I'm restricted to having to record something true and useful, and I have to surrender what could be a valuable piece of information: how much time 52 spent talking to 50. But, as far as billing is concerned, I would save the most valuable thing: that 51 made the call to 50, and that call lasted xxx seconds, no matter who else may or not have spent time in the circuit. Right. And another complication not brought up by this scenario, concerns 3-ways. (really, using assisted xfer, you can form n-way conferences this way)-- CDR's like the 3rd one you listed above, would add up to way more seconds than were actually spent on the call. I guess we could set such CDR's to DOCUMENTATION instead of BILLING (or whatever), to mark them. Haven't yet decided on how I would naturally expect such things to appear in the CDRs. And another issue you brought up earlier-- collect calls. Actually I wrote that later. Maybe the first messages was delayed before showing up on the list. Whatever. I see in the libpri code, that there's a q931 Information element that signals a collect call; perhaps we can insinuate this into the CDR's and dialplan somehow, to either record or even block incoming collect calls. (I guess it'd be a good selling point for moving to PRI.) I had guessed that this is signaled on PRI but wasn't sure. Could be made available to the dialplan as a channel variable. And possibly as a flag or something in the CDRs but that would certainly not pass as a bugfix. Transfers, parking, masquerading, local channels! Bah!!! Humbug!! :) Humbug - I wasn't aware that this word existed in English. Same thing for German. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP usage with Asterisk
Hello, I tried a few months ago to use IMAP with Asterisk; I used either 1.4 or the latest SVN at that time (sorry, don't remember). After a day I had to remove it since Asterisk crashed, mostly in the IMAP client code (the code of UW IMAP). My users wants IMAP back (they loved it) but not in the price of crash... I could not reproduce the crashes at the lab. They only occour on the live system with users. I suggest to write the IMAP client code by the Asterisk developers and not depend on external code. In any case, I'll try this week to upgrade to 1.4.6 version and then add IMAP support and inform what happens. Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best USB Handset and Softphone Combination
On 10/19/07, Mike Clark [EMAIL PROTECTED] wrote: Do they play well with Vista? Hah - I have no idea. We installed Vista on one laptop here when Dell started shipping it. That lasted about 3 days and 10 support tickets from the user. Then we reverted back to XP. Haven't touched Vista since. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2: Incoming calls answered prematurely?
The remote server is where your problem is.
Alan Lord wrote:
Hello,
This message is similar to one I posted before, but with a different
subject line and I've revised the description to hopefully make it clearer.
The basic problem is I am trying to dial 2 numbers simultaneously using
the construct. One device is a locally attached soft SIP phone. The
other device is also a soft SIP phone, but it is on a different Asterisk
server connected over the Internet using IAX. Normal calls between our
servers work fine. However, when I try to dial *both* devices, the
remote Asterisk server answers the call on the IAX channel before it has
checked to see if the real destination device (the SIP phone) is on,
available or busy etc.
This is a problem because I am trying to route calls to a common
voicemail box if both lines are unavailable, busy or go unanswered. Once
the IAX channel answers the incoming call, the dialplan's job is
effectively done. Unfortunately, the remote Asterisk server clears the
call almost immediately, as it finds the real destination extension is
actually not available.
Can anyone see where the problem is? Or suggest a better way?
Many thanks.
Alan
Logs and configuration below:
Here's the last bit of the log (I've edited out the IP address) - we are
both deliberately NOT answering our phones...
Executing [EMAIL PROTECTED]:1] Macro(SIP/101-081d1050,
belllord|SIP/101IAX2/alanb/201|tolc) in new stack
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-081d1050,
SIP/101IAX2/alanb/201|10|tr) in new stack
-- Called 101
-- Called alanb/201
[Oct 17 16:09:47] WARNING[2836]: channel.c:2634 ast_indicate_data:
Unable to handle indication 3 for 'SIP/101-081d1050'
-- SIP/101-081d4fc0 is ringing
-- Call accepted by 80.XXX.XX.XX (format alaw)
-- Format for call is alaw
-- IAX2/alanb-3 answered SIP/101-081d1050
[Oct 17 16:09:47] NOTICE[2836]: cdr.c:434 ast_cdr_free: CDR on channel
'SIP/101-081d4fc0' not posted
[Oct 17 16:09:47] DEBUG[1419]: chan_iax2.c:7435 socket_process:
Immediately destroying 3, having received hangup
[Oct 17 16:09:47] DEBUG[2836]: chan_iax2.c:3176 iax2_hangup: We're
hanging up IAX2/alanb-3 now...
[Oct 17 16:09:47] DEBUG[2836]: chan_iax2.c:3191 iax2_hangup: Really
destroying IAX2/alanb-3 now...
-- Hungup 'IAX2/alanb-3'
== Spawn extension (macro-belllord, s, 1) exited non-zero on
'SIP/101-081d1050' in macro 'belllord'
== Spawn extension (macro-belllord, s, 1) exited non-zero on
'SIP/101-081d1050'
And here's the relevant bits of my extension.conf
[globals]
ALANL=SIP/101 ; My Soft Phone
ALANB=IAX2/alanb/201 ; Alan's Extension
[main_menu] ; Test Dialplan for IVR
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=5) ; Max time between digits
exten = s,n,Set(TIMEOUT(response)=15) ; Max time to wait
exten = s,n,Wait(1)
exten = s,n,Background(welcome-to-bell-lord)
exten = s,n(resume),Background(press-3-for-tolc) ; Short dialogues,
exten = s,n,Background(press-4-for-fondoo) ; rather than one long one
exten = s,n,Background(press-5-for-arrowtees) ; might need to change
exten = s,n,Background(press-6-for-gen-enq) ; frequently.
exten = s,n,WaitExten()
exten = 3,1,Goto(tolc,s,1) ; Dial 3 For The Open Learning Centre
exten = 4,1,Goto(fondoo,s,1) ; Dial 4 for Fondoo Internet
exten = 5,1,Goto(arrowtees,s,1) ; Dial 5 for ArrowTees
exten = 6,1,Goto(gen_enq,s,1) ; For all other enquiries press 6
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(resume)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup() ; Might change this section to go to [gen_enq]
voicemail rather than just hangup.
[tolc]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT}) ; Calls the
belllord Macro with the channel(s) to dial and the current context (for
business voicemail)
[fondoo]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT})
[arrowtees]
exten = s,1,Macro(belllord,${ALANL},${CONTEXT})
[gen_enq]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT})
; Call with Macro(belllord,channel,vmbox)
[macro-belllord] ; Uses macro and DIALSTATUS for local devices
exten = s,1,Dial(${ARG1},10,tr)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED],u) ; business is the
voicemail context, ${ARG2} is the context from which this call came
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED],b)
exten = _s-.,1,Goto(s-NOANSWER,1)
==
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Re: [asterisk-users] [asterisk-biz] DIDX Receives Digium Innovation Award
All of the emails I get from the list have the correct time with the exception of the typical list slowness. All of your emails (and only your emails and spam) are approximately 11 or twelve hours in the future. The email I am responding to has the correct day but the time reads 11:13 PM. I am using Thunderbird 2.0.0.5. If using Outlook, I think the time is correct. Thanks, Steve Rehan Allah Wala wrote: You mean the email that comes from the mailing list or the didx server? IF u can forward the didx email then i can check that Rehan Date sent:Fri, 19 Oct 2007 10:23:00 -0400 From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED], Commercial and Business-Oriented Asterisk Discussion asterisk- [EMAIL PROTECTED] Subject: Re: [asterisk-biz] DIDX Receives Digium Innovation Award Rehan, Pleas fix the time on your email server. I do not need your email sitting the top of my emails for the next 11 hours. It is very annoying. Thanks, Steve Rehan Allah Wala wrote: Super Technologies, Inc.'s DIDXchange has been selected to receive the Digium| asterisk Innovation Award We are thrilled to be honored with this award and want to thank all of you and all of the judges and all of them at Digium for Making such a Great Product Asterisk available to all of the world and letting Companies like our use it for our Innovations. Digium will send a press release the week of October 22nd as well as announcing the winners during a presentation at Digium|Asterisk World in Boston, Massachusetts during Fall VON Oct 30 - Nov 1, 2007. We, that is each of you the DIDXchange members and we, all of our DIDX care team members, share in this success. It only makes us try even more than ever to help you be most effective and successful regarding your DID needs. For more information on the award Check out _http://www.digium.com/en/company/awards/innovation.php_ . We hope you can visit us at the Fall Von and Digium|Asterisk World 2007 this year and be a part of this great event with us, as without you, it would not have been possible. Visit _http://www.didx.net/fallvon2007_and Even if you cannot attend, there are many benefits for registering, so please don't miss it and at least sign up now! DIDXchange will be at booth 1263. for more information viit www.didx.net Rehan Ahmed AllahWala Msn/Yahoo/GoogleTalk/Email: [EMAIL PROTECTED] http://www.supertec.com/ - Internet Telephony Solutions Http://www.DIDX.net - DID Number Market Place. Don't Remember Me ? Visit http://www.Rehan.com ~~~ First they ignore you, then they laugh at you, then they fight you, then you win. By Gandhi. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz Rehan Ahmed AllahWala Msn/Yahoo/GoogleTalk/Email: [EMAIL PROTECTED] http://www.supertec.com/ - Internet Telephony Solutions Http://www.DIDX.net - DID Number Market Place. Don't Remember Me ? Visit http://www.Rehan.com ~~~ First they ignore you, then they laugh at you, then they fight you, then you win. By Gandhi. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2: Incoming calls answered prematurely?
Hello,
This message is similar to one I posted before, but with a different
subject line and I've revised the description to hopefully make it clearer.
The basic problem is I am trying to dial 2 numbers simultaneously using
the construct. One device is a locally attached soft SIP phone. The
other device is also a soft SIP phone, but it is on a different Asterisk
server connected over the Internet using IAX. Normal calls between our
servers work fine. However, when I try to dial *both* devices, the
remote Asterisk server answers the call on the IAX channel before it has
checked to see if the real destination device (the SIP phone) is on,
available or busy etc.
This is a problem because I am trying to route calls to a common
voicemail box if both lines are unavailable, busy or go unanswered. Once
the IAX channel answers the incoming call, the dialplan's job is
effectively done. Unfortunately, the remote Asterisk server clears the
call almost immediately, as it finds the real destination extension is
actually not available.
Can anyone see where the problem is? Or suggest a better way?
Many thanks.
Alan
Logs and configuration below:
Here's the last bit of the log (I've edited out the IP address) - we are
both deliberately NOT answering our phones...
Executing [EMAIL PROTECTED]:1] Macro(SIP/101-081d1050,
belllord|SIP/101IAX2/alanb/201|tolc) in new stack
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/101-081d1050,
SIP/101IAX2/alanb/201|10|tr) in new stack
-- Called 101
-- Called alanb/201
[Oct 17 16:09:47] WARNING[2836]: channel.c:2634 ast_indicate_data:
Unable to handle indication 3 for 'SIP/101-081d1050'
-- SIP/101-081d4fc0 is ringing
-- Call accepted by 80.XXX.XX.XX (format alaw)
-- Format for call is alaw
-- IAX2/alanb-3 answered SIP/101-081d1050
[Oct 17 16:09:47] NOTICE[2836]: cdr.c:434 ast_cdr_free: CDR on channel
'SIP/101-081d4fc0' not posted
[Oct 17 16:09:47] DEBUG[1419]: chan_iax2.c:7435 socket_process:
Immediately destroying 3, having received hangup
[Oct 17 16:09:47] DEBUG[2836]: chan_iax2.c:3176 iax2_hangup: We're
hanging up IAX2/alanb-3 now...
[Oct 17 16:09:47] DEBUG[2836]: chan_iax2.c:3191 iax2_hangup: Really
destroying IAX2/alanb-3 now...
-- Hungup 'IAX2/alanb-3'
== Spawn extension (macro-belllord, s, 1) exited non-zero on
'SIP/101-081d1050' in macro 'belllord'
== Spawn extension (macro-belllord, s, 1) exited non-zero on
'SIP/101-081d1050'
And here's the relevant bits of my extension.conf
[globals]
ALANL=SIP/101 ; My Soft Phone
ALANB=IAX2/alanb/201 ; Alan's Extension
[main_menu] ; Test Dialplan for IVR
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=5) ; Max time between digits
exten = s,n,Set(TIMEOUT(response)=15) ; Max time to wait
exten = s,n,Wait(1)
exten = s,n,Background(welcome-to-bell-lord)
exten = s,n(resume),Background(press-3-for-tolc) ; Short dialogues,
exten = s,n,Background(press-4-for-fondoo) ; rather than one long one
exten = s,n,Background(press-5-for-arrowtees) ; might need to change
exten = s,n,Background(press-6-for-gen-enq) ; frequently.
exten = s,n,WaitExten()
exten = 3,1,Goto(tolc,s,1) ; Dial 3 For The Open Learning Centre
exten = 4,1,Goto(fondoo,s,1) ; Dial 4 for Fondoo Internet
exten = 5,1,Goto(arrowtees,s,1) ; Dial 5 for ArrowTees
exten = 6,1,Goto(gen_enq,s,1) ; For all other enquiries press 6
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(resume)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup() ; Might change this section to go to [gen_enq]
voicemail rather than just hangup.
[tolc]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT}) ; Calls the
belllord Macro with the channel(s) to dial and the current context (for
business voicemail)
[fondoo]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT})
[arrowtees]
exten = s,1,Macro(belllord,${ALANL},${CONTEXT})
[gen_enq]
exten = s,1,Macro(belllord,${ALANL}${ALANB},${CONTEXT})
; Call with Macro(belllord,channel,vmbox)
[macro-belllord] ; Uses macro and DIALSTATUS for local devices
exten = s,1,Dial(${ARG1},10,tr)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED],u) ; business is the
voicemail context, ${ARG2} is the context from which this call came
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED],b)
exten = _s-.,1,Goto(s-NOANSWER,1)
==
--
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Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
shadowym wrote: Or your could use a touch screen with Flash Operator Panel. Just a suggestion out of left field. snipped a bunch shadowym: Do you have a specific setup w/touchscreen that you have deployed and that works well? Thanks, Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut()
ResponseTimeout was deprecated in 1.2 and removed in 1.4. Was this information not in the upgrade.txt file in 1.2 and 1.4? bilal ghayyad wrote: Hi List; My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup To what this related? About my extensions.conf file, I set priorityjumpin = yes and I set autofallthrough = no (and I am sure it is not related to the problem with ResponseTimeout application). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ResponseTimeOut()
bilal ghayyad wrote: My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup To what this related? There is no ResponseTimeout() in 1.4. Use Set(TIMEOUT(response)=10) core show function TIMEOUT And have a look at core show application WaitExten Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX
[EMAIL PROTECTED] wrote: Hi, I'm running some Asterisk-machines, and on one of them i get this errors in the CLI, but i don't know what that means. Hardware: Digium 4-Port E1 Card with HWEC Intel Pentium D 3 GHz 2 GB RAM SATA Harddisk Supermicro Mainboard Software: latest libpri/zaptel/asterisk of version 1.2 I tried also asterisk version 1.4.x, but there the problem occurs every 10 calls, on asterisk 1.2 its about every 100 calls. Did this recently start, like after you upgraded or is this something that has always been a problem for you since you installed? If it has always been a problem, can you post a `pri debug span x` trace of a call when this happens? That will help to know more about what is going on here. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
On Wed, 2007-10-17 at 02:41 +0200, Philipp Kempgen wrote: Steve Murphy wrote: It's not a bad idea, but I think the philosophy would be that whatever turns CDR records into billing statements could/should/would decide which to skip, and which to process, and not something in Asterisk. It's easier just to state what happens, and let each org that uses the data decide what to do with it. I like that approach. I think it should be possible to tell exactly what happened just by looking at the CDRs (e.g. who transferred which call to whom etc.) As far as picking up handsets, and dropping them again, such would always have 0 duration and billsec figures, because an answer is needed for either field to be greater than zero. I guess in this regime, they'd all be dropped if you set your threshold at 1 or more... Dropping records with duration == 0 is an easy task for custom post-processing. Even more so if you store your CDRs in an SQL database. For instance, if Zap/52 dials Zap/51, 52 --- dials talks --- 51 who hookflashes and dials Zap/50, 51 --- dials --- 50 and 51 hangs up, leaving 52 and 50 to talk away, 52 --- talks --- 50 we should get 1 cdr that records the call from 52 to 51, which would last until the hookflash; No doubt about it. and a second CDR that would be from 51 to 50, which would start at either chan 50/51 channel creation time, or even at hookflash time, have an answer time when 50 picked up, and last until either 50 or 52 hang up. Right. But why should it start at 50-51 channel creation time? That way you would think (by looking at the CDRs) that 51 talked to 50 for longer than they did. I'd prefer hookflash time. Well, the start time isn't as important as the answer time; because your billing times from answer to end. The time from start to answer is how much time it took to dial, wait, and have the call answered... which people usually don't pay as much attention to. 50's channel creation time will be when 50 picked up the phone to answer the call from 51. 51's channel creation time will be when 51 picked up the phone to answer the call from 52. If we use 51's channel creation time as the start time, it would be possible to see that 52's conversation with 51 and 51's with 50, overlap. It may not help much, but it's a hint that 52 was there. Even tho 52 and 50 might talk an hour, 51 is the one who dialed, and therefore seems naturally responsible for the call... 51 is responsible, correct. But the fact that it was 52 (not 51) who talked to 50 might be equally important. True. How about splitting the src into rsp (who's responsible for the call, i.e. who should pay the bill) and src (who was involved in the audio bridge)? Example: rsp src dst duration billsec 5252 51 130 120 5151 50105 5152 50 3610 3600 Actually, I've been thinking about this; adding a CDR field to record the responsible party for a call is a good way to handle these situations. But, in 1.4, I really can't add a new CDR field and call it a 'bug fix'. It really is a 'new', 'enhanced' sort of thing. So, this kind of change will have to go into trunk at the moment. Since I can't add the new field into 1.4, I'm restricted to having to record something true and useful, and I have to surrender what could be a valuable piece of information: how much time 52 spent talking to 50. But, as far as billing is concerned, I would save the most valuable thing: that 51 made the call to 50, and that call lasted xxx seconds, no matter who else may or not have spent time in the circuit. And another complication not brought up by this scenario, concerns 3-ways. (really, using assisted xfer, you can form n-way conferences this way)-- CDR's like the 3rd one you listed above, would add up to way more seconds than were actually spent on the call. I guess we could set such CDR's to DOCUMENTATION instead of BILLING (or whatever), to mark them. And another issue you brought up earlier-- collect calls. I see in the libpri code, that there's a q931 Information element that signals a collect call; perhaps we can insinuate this into the CDR's and dialplan somehow, to either record or even block incoming collect calls. (I guess it'd be a good selling point for moving to PRI.) You'd not believe how tricky getting these sequences to generate the right CDR data can be! It's almost humorous! This could be much easier if Asterisk did not have fancy features like transfers etc. ;) I totally agree Transfers, parking, masquerading, local channels! Bah!!! Humbug!! :) Regards, Philipp Kempgen smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
Perssy Llamosas wrote: I doubt it. hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/ I think that is the sort of thing the OP would classify as religious grounds. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP Trunk, but need to register on the destination as gatekeeper client
Hi List; I need to do IP Trunk between Asterisk and another softswitch provider, the softswitch support SIP but requires Asterisk to register for this IP Trunk (it should appears as gatekeeper entity that does registeration to another gatekeeper entity). How can I configure this SIP trunk to do registeration with tht softswitch, so I can send the calls for it? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Background not listening?
Any chance that your dtmf is not set up correctly ?
- Original Message -
From: Michael Munger
To: asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 10:30 PM
Subject: [asterisk-users] Background not listening?
This ridiculously simple IVR is not listening to dial tones to dial an
extension. I can hit the extension all I want, and nothing happens. Just DTMF
in my ear.
I need another pair of eyes to tell me what I am missing here. Anyone see a
mistake?
[ivr]
exten = s,1,Answer()
exten = s,n,Background(tempivr)
exten = s,n,WaitExten(10)
exten = s,n,Goto(inbound,5250,1) ; Run this back to inbound context as if
the call was being re-originated.
exten = s,n,Hangup()
exten = _,1,Macro(dial-ext|${EXTEN})
Yours,
Michael
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Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
On my SPA3102 on the Admin Advanced SIP page: Subsitute VIA Addr: yes Send Resp To Src Port: yes I also set the RTP Port Min RTP Port Max so that my NAT router could be set up to forward RTP packets to this device. This is quite a good posting about setting up Linksys devices to handle NAT (talks about voxalot service but general advice is good) : http://forum.voxalot.com/voxalot-general/1091-voxalot-sipura-ata-tutorial-comprehensive-walkthrough.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Per Jessen Sent: Friday, October 19, 2007 6:01 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] A linksys SPA921 behind NAT and firewall Per Jessen wrote: [EMAIL PROTECTED] wrote: Did you set NAT Keep Alive Enable: = Yes for the line in question in the SPA's configuration? Uh, no, not specifically and I'm guessing it's not set by default? The SPA921 config has a NAT Keep Alive Intvl which is set to 15 by default, which I'm taking to mean it has NAT keep alives enabled. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Howto get origin IP address from SIP call reliably
Hi, incoming SIP calls have a channel name in the form of: SIP/ip-adresss-of-peer-handle This is a way to get fetch the IP address of the remote side of a SIP call - in most cases. However, sometimes, instead of the IP address, there is a host name in the channel name. I assume, this value in the channel name is not the real IP address, but just a field filled in by the remote SIP client. Thus, this is not a reliable way to check the origin of a SIP call. What is a reliable way to read the real IP address of the origin of a SIP call? Regards, Roger. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 bit asterisk
Tzafrir Cohen wrote: By now there are quite a few x86_64 Asterisk users that complain if something breaks. Been using it on a 64-bit P4 with debian 4.0/1 (amd64) for some time now without a hitch. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
Per Jessen wrote: [EMAIL PROTECTED] wrote: Did you set NAT Keep Alive Enable: = Yes for the line in question in the SPA's configuration? Uh, no, not specifically and I'm guessing it's not set by default? The SPA921 config has a NAT Keep Alive Intvl which is set to 15 by default, which I'm taking to mean it has NAT keep alives enabled. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 bit asterisk
On 10/19/07, Tzafrir Cohen wrote: On Thu, Oct 18, 2007 at 11:24:24PM -0400, Baji Panchumarti wrote: I hope you have better success than I did, my problem was not so much with asterisk in particular but 64-bit in general. Examples of problems using CentOS 4.5 on x86_64 - many problems loading php5 mysql from package repositories. What repositories did you use? I don't recall CentOS 4.5 including PHP5. Is this a third-party package? If so: stick with the official PHP4 packages, or complain to whoever packaged those PHP5 packages. correct, PHP5 is not included, centosplus repository - a few asterisk functions don't work, eg STRFTIME() What version of Asterisk? What bug number in bugs.digium.com ? 1.4.12 -- ( copy pasting from a previous thread ) -- -- 2007-09-12 20:12 + [r82285] Tilghman Lesher [EMAIL PROTECTED] -- -- * main/stdtime/private.h, main/stdtime/tzfile.h, --include/asterisk/localtime.h, main/stdtime/localtime.c: Working --on issue #10531 exposed a rather nasty 64-bit issue on --ast_mktime, so we updated the localtime.c file from source. --Next we'll have to write ast_strptime to match. -- -- 1.4.12 changelog -- http://svn.digium.com/view/asterisk/tags/1.4.12/ChangeLog?view=markup Perhaps the distro you are using is more caught up on 64 bit. Debian has long ago included Asterisk on x86_64 and other platforms. And it works, as one of the packagers actually has had a x86_64 for quite some time. I'll get debian a try in the near future since I hear you praise it often over other distros. thnx, -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64 bit asterisk
On Thu, Oct 18, 2007 at 11:24:24PM -0400, Baji Panchumarti wrote: I hope you have better success than I did, my problem was not so much with asterisk in particular but 64-bit in general. Examples of problems using CentOS 4.5 on x86_64 - many problems loading php5 mysql from package repositories. What repositories did you use? I don't recall CentOS 4.5 including PHP5. Is this a third-party package? If so: stick with the official PHP4 packages, or complain to whoever packaged those PHP5 packages. If those are the official packages, could you please give a bug number in bugzilla.redhat.com or in CentOS's bug tracker? - a few asterisk functions don't work, eg STRFTIME() What version of Asterisk? What bug number in bugs.digium.com ? Perhaps the distro you are using is more caught up on 64 bit. Debian has long ago included Asterisk on x86_64 and other platforms. And it works, as one of the packagers actually has had a x86_64 for quite some time. Everything upgraded/updated without a hitch on 32 bit. 64 bit is a no go unless you are running packages that have matured for atleast a couple of years old...imho. By now there are quite a few x86_64 Asterisk users that complain if something breaks. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free help
On Fri, Oct 19, 2007 at 01:40:20AM +, Rony Ron wrote: Hello all, i would like to have references so i'm giving free help for any project (commercial or public). One useful and obvious reference: http://www.catb.org/~esr/faqs/smart-questions.html -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Receptionists Phone suggestions? (Not Snom370)
Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details below if you/re interested). I've verified this problem with Snom who's response is that the receptionist should answer all of the incoming calls before trying to do a transfer - That's just Bonkers! So... any suggestions? Details of Snom 370 problem for the record: Snom370 gets a Call (Call A). Snom370 answers Call A. Call A wants to be transferred to Phone C. Snom370 has another call ringing (Call B). Snom370 presses HOLD button gets Dialtone. Call A is on Hold, Call B still ringing. Snom370 Dials Phone C (Call C). Snom370 talks to Call C. Snom370 presses TRANSFER. The display shows: CallA CallB The soft keys now show and . Pressing them does nothing. When the TRANSFER button is pressed again, CallA is connected to CallB (the original caller is now talking to the previously unanswered party) not what one wanted to happen! It's not difficult to see why my client is throwing their toys out of the pram and I'm going to have to replace the Snoms at my expense :-( -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2: Calls answered before extension is tested?
[EMAIL PROTECTED] wrote: So your problem is: -- IAX2/alanb-3 answered SIP/101-081d1050 Except the remote end didn't actually answer the call? The problem is your remote end... its answering the call. All the IAX hardphones I've seen don't seem to be the highest of quality honestly. Hi, That's not a phone. That is another Asterisk server, configured with an IAX2 - IAX2 connection between our two offices. His real extension is a Twinkle Softphone. This is what I was questioning initially. It appears as though asterisk is answering the incoming IAX2 connection call *before* actually checking if the true destination is actually available or not. Thanks for the input - I probably didn't explain myself clearly enough. Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free help
Doug wrote: At 20:40 10/18/2007, Rony Ron wrote: Hello all, i would like to have references so i'm giving free help for any project (commercial or public). regards, Can you come over and wash my car? I could write you a script to wash your car. You'd just need some kind of interface to do the mechanical part of the work. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and wall displays/reader boards
Has anyone used an LED wall display with asterisk? I have a customer who has an ancient telecorp system that drives an LED wall display. It shows the number of agents signed in, calls in queue, hold time, etc. It also sounds an alarm if the hold time exceeds a set value. I'm looking to use asterisk to replace the telecorp system. I know it can do all the CDR and historical data, but I haven't found anything on this. The current display is currently connected via serial (rj-11) but I would be open to getting a newer board with IP connectivity. thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free help
At 20:40 10/18/2007, Rony Ron wrote: Hello all, i would like to have references so i'm giving free help for any project (commercial or public). regards, Can you come over and wash my car? -- Your next Partner ! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sorta OT: Bounty for Click to Call plugin for IE
There is a free dialer from http://www.snapanumber.com/ If I remember correctly, it will let you click on phone numbers in web pages. -- -- Steven http://www.glimasoutheast.org Michael Graves [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I'm in process of transitioning a number of offices to a hosted virtual pbx from Junction Networks. It's a combination of OpenSER and Asterisk. They have a nice click-to-call extension for Firefox, but I need the equivalent for IE so that it can work with our CRM system. Junction told me that they have a bounty on offer for this if someone's interested in doing the work. Would the availability of the Firefox code make it easier to do an ActiveX implementation? Any takers? Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2: Incoming calls answered prematurely?
Eric ManxPower Wieling wrote:
Alan Lord wrote:
Eric ManxPower Wieling wrote:
The remote server is where your problem is.
Thanks for the reply but I can call the extension in question normally
and it works fine. The problem is that the IAX trunk appears to be
answering before it knows if the physical destination is available or
not. I have read through every option I can find on IAX and elsewhere
and I can't see how this functionality can be changed or influenced.
How do you know that the far end is not answering and then providing an
ringing tone. Asterisk does not magically answer IAX calls. Playback
and Background as well as other apps will answer the line unless told
not to.
When I tried this test today, I know the far end wasn't answering
because my colleague, his computer and his SIP phone were not there. So
there is no way that that call should have been answered.
His extension definition is:
[internal]
exten=201,1,Dial(${ALANB},10)
exten=201,2,VoiceMail(u201)
exten=201,3,Hangup()
The call was cleared down almost as soon as it was answered so I am
unclear as to why this occurred.
Thanks
Alan
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Re: [asterisk-users] Best USB Handset and Softphone Combination
Erik Anderson wrote: On 10/19/07, Steve Totaro [EMAIL PROTECTED] wrote: Any advice on softphones, handsets, or practical experience with this sort of deployment? It would be very nice if there was a central way of provisioning the phones. I've deployed several setups internally using X-Lite and these headsets: http://www.newegg.com/Product/Product.aspx?Item=N82E16826275009 Haven't heard of a single problem thus far. -erik Erik: Do they play well with Vista? Mike Clark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Live Conference about asterisk and voip: reminder 12:30 PM EDT Friday
for those of you who have not joined the conference call yet, I highly recommend it. there is always several interesting tidbits that will help you in your * implementations... see you at 12:30p today! daveC randulo wrote: As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Glare on Incoming Calls
How I change my configuration to reduce this issue. I have this setting on
my zapata.conf
signalling=fxs_ks
group=1
callgroup=1
pickupgroup=1
channel=1
signalling=fxs_ks
group=2
callgroup=1
pickupgroup=1
channel=2;
singalling=fxs_ks
group=3
callgroup=1
pickupgroup=1
channel=3;
singalling=fxs_ks
group=4
callgroup=1
pickupgroup=1
channel=4
and for outbound calls I have this context on my extensions.conf
[out-callb]
exten = 44,1,Set(LANGUAGE()=es)
exten = 44,n,ChanIsAvail(Zap/g1Zap/g2Zap/g3Zap/g4)
exten = 44,n,GotoIf($[${AVAILCHAN} = ]?4:6)
exten = 44,n,Congestion
exten = 44,n,Hangup
exten = 44,n,Playback,ggestion/varios/moment
exten = 44,n,SetMusicOnhold(dialtone)
exten = 44,n,Set(TIMEOUT(response)=10)
exten = 44,n,Set(TIMEOUT(digit)=5)
exten = 44,n,WaitExten(25|m(dialtone))
Date: Thu, 18 Oct 2007 17:07:03 -0400
From: C F [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Incoming calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Glare that's what it's called, if the number you advertise as your
business number is zap/1 then use zap/G1 to dial out, otherwise use
zap/g1 to dial out. This will reduce but not eliminate the problem.
On 10/18/07, Gustavo Gonzalez [EMAIL PROTECTED] wrote:
Hello I have a question about incoming calls on TDM400P cards. I want to
know why an incoming call appear in a sorpresive way on a phone that I
pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
channels )that are free. I have four lines connected to my TDM400P card
and
when I get a free Zap channel to call I hear the voice of a people on the
other side from an incomming call, I think that asterisk bridge my free
channel with incomming calls but how do this?Thanks for any idea.
Alejandro González
Grupo Gestión
4384-0660
www.grupo-gestion.com.ar
[EMAIL PROTECTED]
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Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
On Fri, 2007-10-19 at 09:12 +0100, Russell Brown wrote: Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? Personally I'm happy with a Linksys SPA-962 + the 932 sidecar. With the latest firmware, the Busy Lamp Fields work well, and can also be used as Speed Dial buttons at the same time. I've also heard good reports from people using Polycom and Aastra phones. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and wall displays/reader boards
QueueMetrics is able to prepare a realtime screen meant for a video projector or large LCD screen to display to show call-center stats in real-time. We have quite a number of customers who used old linux boxes connected to the right display that just start up, start firefox and go to a specific url. They seem to like it - better than LCD stripes in any case :) l. On Fri, 19 Oct 2007 08:28:52 +0200, o o [EMAIL PROTECTED] wrote: Has anyone used an LED wall display with asterisk? I have a customer who has an ancient telecorp system that drives an LED wall display. It shows the number of agents signed in, calls in queue, hold time, etc. It also sounds an alarm if the hold time exceeds a set value. I'm looking to use asterisk to replace the telecorp system. I know it can do all the CDR and historical data, but I haven't found anything on this. The current display is currently connected via serial (rj-11) but I would be open to getting a newer board with IP connectivity. thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX
Hi, I'm running some Asterisk-machines, and on one of them i get this errors in the CLI, but i don't know what that means. Hardware: Digium 4-Port E1 Card with HWEC Intel Pentium D 3 GHz 2 GB RAM SATA Harddisk Supermicro Mainboard Software: latest libpri/zaptel/asterisk of version 1.2 I tried also asterisk version 1.4.x, but there the problem occurs every 10 calls, on asterisk 1.2 its about every 100 calls. any ideas on this? Thanks a lot Nico ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
Russell Brown wrote: Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Linksys SPA94x/6x perhaps. I don't know if it has the transfer problem or not. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BBC on Atserix
Hi all, Well, actually, I'm looking at asterisk from the development/SIP side of things, not the cartoons. Or that's what I hope my project leader wants me to do... Best regards, Matti Zemack, BBC RD, Kingswood Warren, UK http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
[EMAIL PROTECTED] wrote: Did you set NAT Keep Alive Enable: = Yes for the line in question in the SPA's configuration? Uh, no, not specifically and I'm guessing it's not set by default? thanks. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and wall displays/reader boards
o o wrote: Has anyone used an LED wall display with asterisk? I have a customer who has an ancient telecorp system that drives an LED wall display. It shows the number of agents signed in, calls in queue, hold time, etc. It also sounds an alarm if the hold time exceeds a set value. I'm looking to use asterisk to replace the telecorp system. I know it can do all the CDR and historical data, but I haven't found anything on this. The current display is currently connected via serial (rj-11) but I would be open to getting a newer board with IP connectivity. Use a web server with some dynamic pages. You could either do some fancy Ajax stuff or the old method of reloading the page every x seconds. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My spa has a mind of its own
I had a similar issue a while ago. Check your dial plan. Are you forwarding to your cell phone's V-Mail as fallback? I had the issue where I was getting callbacks from asterisk if one phone was on DnD and the calll wasn't answered. Becarefull of your dial() commands and the delays you use. Steve Edwards wrote: I have a Sipura SPA-841. It's developed a nasty habit. At random times, it likes to dial my cell phone voicemail number and play my messages to anybody who happens to be within earshot. Any clues where to look at what's going on? My voice mail number (extension 220 in my dialplan) is the only number being dialed. When this happens, show channels looks like this: IAX2/NuFone-1(None) Up Bridged Call(SIP/spa841-09f083 SIP/spa841-09f08388 [EMAIL PROTECTED]:5 Up Dial(IAX2/mumble:mumble which looks the same as if I dial it myself. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- It is completely one's own responsibility to think outside the cucumber. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Poll: Asterisk IMAP feedback (was: Is anyone successfully using IMAP storage)
Hello, Are you using Asterisk 1.4 ? If positive, are you then successfully using IMAP storage ? Your input would be very valuable to decide if rewite of IMAP storage could be considered as bug fix (non one uses IMAP now) or as a new feature (many use IMAP storage today). So please, take a few seconds to reply as up to now (4 answers), successful IMAP user share = 0% ! Regards PS: If someone has a more effective way to gather user feedback, do not hesitate to tell. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best USB Handset and Softphone Combination
On 10/19/07, Steve Totaro [EMAIL PROTECTED] wrote: Any advice on softphones, handsets, or practical experience with this sort of deployment? It would be very nice if there was a central way of provisioning the phones. I've deployed several setups internally using X-Lite and these headsets: http://www.newegg.com/Product/Product.aspx?Item=N82E16826275009 Haven't heard of a single problem thus far. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom ip330/ip501 second ethernet port
Kevin Smith wrote: Hi Robert, While I'm not sure how our network compares with yours, we run about twenty 601 phones along with our office workstations (some stations are without a phone). Each station with a phone is connected with the other Ethernet port on the phone so we have one drop to each station. The phones are on a separate VLAN from the rest of the network as well. From the user end, I have not had a report of any problems with the connections, call quality, etc. I would say give it a shot, maybe with a larger network that could change, but for a small office like I'm in charge of, it is working just fine. The major issue with this is most pc's are now coming with gigabit ethernet connections. Going to gigabit speeds is such a huge improvement it's often worth the extra expense to add a second drop to each location. Profiles will load faster, Outlook-exchange interactions work much cleaner. When gigabit capable phones are more prevalent, this becomes a non-issue. Right now, there are very few gigabit phones and none that are affordable. Robert McNaught wrote: Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
I think most phones somehow have this kind of behaviour : transfer button applies to ongoing call and so on. What happens if you don't press TRANSFER again (when display shows Call A CallB) ? Have you tried call parking ? What if you used blind transfer instead ? If receptionist is busy, assisted transfer might be confusing under pressure. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is anyone successfully using IMAP storage
Hello, I think I will create a new thread with a more attractive tittle to gather more feedback, so that decisions are easier. Up to now (4 answers): no one uses IMAP storage. As 1.4 is not so widely spread, chances are this IMAP feature is not used at all. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hide passwords in SIP.conf
On Fri, Oct 19, 2007 at 11:15:35PM +0100, Alan Lord wrote: Frederico Madeira wrote: Hi guys, There is other way instead plain text to define passwords in sip.conf ? In register, peers and extensions ? Thanks. Depending on how your asterisk server is setup to run, if you chmod /etc/asterisk as 750 and the files underneath as 640, then only the user and group owner can read (+ only owner user can write). Others will not even see the existence of the directory or files... My server runs as user asterisk and group asterisk. But then again, someone with the permission to connect to the control socket or to the manager interface (with the command write permission) can also issue sip show users, regardless of where you actually keep the passwords. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
